Archives : July-2018
Sorry, I see I have submitted a testing version of the dialplan fragment. Actual extension 7777 is: ;listen to recording exten=>7777,1,Answer() exten=>7777,n,NoOp(Requesting File ${recfile}) exten=>7777,n,NoOp(Rec file set to ${recfile}) exten=>7777,n,NoOp(..
Guys I have the following dialplan code that I use to play back recordings, the filename being provided in an originate statement to the AMI (AJAM) interface: Action: Originate ActionID: test Channel: SIP/3015 Exten: 7777 Context: local Priority: 1 Caller..
Could you give us more details what you are trying to do ?Your request seems special for m..
Who do you want to whisper to? Both channels? The way it works though is snoop channel creates a channel that you can treat like any other (add to a bridge, record, etc) – audio coming from it is a copy of the audio from the channel you are snoop..
Maybe Digium should include a G729 codec inside Asterisk. What is keeping them from..
I am sure Digium would not prefer to acknowledge this, but the phenomenal growth of Asterisk is due to the aavailability of a free G729 codec compiled and distributed free by Arkad..
I just finished installing a brand new server with CentOS 7.5 and Asterisk 13.22.0 and the minute I a call from the PSTN (from a SIP trunk) bridges with any SIP phone Asterisk crashes: Jul 20 10:59:53 localhost kernel: asterisk[2819]: segfa..
I suspect you’re encountering behavior that is working as intended.Normally, when Asterisk plays back a file, it scans the file system for all files with the provided sound file name. For each file that it finds with a given file extension, it pi..
I am having one of those days. We just replaced an old Asterisk 1.8 server with a new Asterisk 13.21.1 (both using Freepbx) and almost everything is working except for some incoming calls directed to a Cisco SPA-8000. The PSTN trunk is SIP..
HelloIm currently using Asterisk 13 with the chan_sip sip driver. The extensions are offloaded via realtime module to a MySQL database (via ODBC). So basically I have a MySQL Table with the SIP users + SIP passwords and the other stuff from the stand..